The present invention relates to network analysis, and more particularly to analyzing Voice over Internet Protocol (VoIP) calls.
Voice signals are transmitted over a packet network by first formatting the voice signal data stream into multiple discrete packets. In a Voice Over Internet Protocol (VoIP) call, an originating voice gateway quantizes an input audio stream into packets that are placed onto a packet network and routed to a destination voice gateway. The destination voice gateway decodes the packets back into a continuous digital audio stream that resembles the input audio stream. As an option, a compression or decompression algorithm may be used on the quantized digital audio stream to reduce the communication bandwidth required for transmitting the audio packets over the network.
Similar to conventional Internet Protocol, VoIP includes a plurality of layers. Prior Art FIG. 1 illustrates a plurality of exemplary well known layers 10 associated with VoIP. As shown, such layers include at least one application layer 12 and a plurality of session layers 14 positioned below the application layer 12. While not shown, at least one connection layer may be positioned below the session layer. By way of example, the application layer 12 may include H.323. H.323 is a standard approved by the International Telecommunication Union (ITU) in 1996 to promote compatibility in videoconference transmissions over IP networks. Further included as session layers are H.225.0, H.245, real-time transport protocol (RTP), and real-time transport control protocol (RTCP). It should be noted that VoIP calls can employ various protocols for communication purposes.
The Quality of Service (QoS) of VoIP calls can degrade due to congestion on the packet network or failure of network processing nodes in the packet network. Quality of service can include anything from call sound quality to the ability and responsiveness of the VoIP network in establishing new VoIP calls. IP network reliability has not been proven to be in the same class as a traditional switched Public Services Telephone Network (PSTN).
Due to a need to understand, troubleshoot and optimize a particular network to improve VoIP calls, there is an on-going desire for traditional network assessment tools to be tailored to monitor network parameters specific to VoIP calls. Network assessment tools referred to as xe2x80x9canalyzersxe2x80x9d are often relied upon to analyze networks communications at a plurality of layers. One example of such analyzers is the SNIFFER ANALYZER(trademark) device manufactured by NETWORK ASSOCIATES, INC(trademark). Analyzers have similar objectives such as determining why network performance is slow, understanding the specifics about excessive traffic, and/or gaining visibility into various parts of the network.
As mentioned earlier, network analyzers collect information at a plurality of layers. Each set of layer-specific data is conventionally stored in a buffer xe2x80x9cobjectxe2x80x9d by the network analyzer. In particular, a session object, an application object, etc. are each used to store network traffic information at session and application layers, respectively. Further, each specific type of voice protocol data may be stored in a dedicated object.
In use, a separate object is established for data collected at each application and session layer for each VoIP call. With the number of such objects growing proportionally with the overall VoIP calls and the number of voice protocols associated therewith, managing related network data for monitoring, reporting and analysis purposes may become quite cumbersome.
There is thus a need for a more efficient and effective technique for collecting and managing VoIP call network data for analysis purposes.
A system, method and computer program product are provided for filtering various voice protocols. A plurality of voice protocols is initially displayed. Next, an indication is received from a user as to the selection of the voice protocols. It is further determined as to a particular filtering mode that is currently operating. Next, the selected voice protocols are filtered in the determined filtering mode.
In one embodiment, the voice protocols may include H.323, H.225, H.245, registration admission status protocol (RAS), real-time transport protocol (RTP), real-time transport control protocol (RTCP), session description protocol (SDP), session announcement protocol (SAP), session initiation protocol (SIP), skinny client control protocol (SCCP), and/or media gateway control protocol (MGCP). Further, the filtering modes may include a monitor mode, a capture mode, and/or a display mode.
In another embodiment, the voice protocols may be displayed in response to the selection of a filter icon. Moreover, the voice protocols may be selected utilizing a plurality of check boxes. Still yet, the filtering may be displayed in a tree representation for reporting purposes.
Another system, method and computer program product are provided for filtering various voice protocols. Initially, a plurality of voice protocol filters is displayed. An indication is then received from a user as to the selection of at least one of the voice protocol filters. Next, a plurality of voice protocols associated with the selected voice protocol filter is displayed. Another indication from the user is received as to the selection of the voice protocols to be associated with the voice protocol filter.
During an analysis, a voice application call is then identified, and a plurality of connection, session, and application objects associated with the voice application call are collected based on the selected voice protocols. Such connection objects, session objects, and application objects are subsequently displayed. By allowing a user to select the voice protocol filters and further define the same, the collection and analysis of the connection, session, and application objects is more focused and manageable.